Please post a full SIP trace with console output. It looks like you have bypass_media turned on. More than likely that broken reply SDP is coming from another device, not FreeSWITCH. On Wed, Mar 14, 2012 at 2:40 AM, Faisal Rehman <faisal.rehman22@hotmail.com> wrote:
Hi Everyone,
We are working on scenario where Freeswitch is terminating calls a vendor with g729 passthru mode. But when we get the call terminated , termination side replies with static payload but freeswitch replies with Dynamic payload. Our customer wants to be replied with only static payload. How can we solve this?
Situation:
Originator -> freeswitch -> termination
In session progress :
Termination side sends,
m=audio 64034 RTP/AVP 18 101^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M
Freeswitch sends following to origination.
m=audio 0 RTP/AVP 96 101 a=rtpmap:96 G729/8000 a=fmtp:96 annexb=no
Configuration File
external.xml:
<settings> <param name="debug" value="0"/> <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. --> <!-- <param name="shutdown-on-fail" value="true"/> --> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public2"/> <param name="dtmf-duration" value="2000"/> <!-- <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> --> <!-- <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> --> <param name="disable-transcoding" value="true"/> <param name="inbound-late-negotiation" value="false"/> <param name="inbound-bypass-media" value="true"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <!--<param name="enable-100rel" value="true"/>--> <!-- This could be set to "passive" --> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <!-- used to share presence info across sofia profiles manage-presence needs to be set to passive on this profile if you want it to behave as if it were the internal profile for presence. --> <!-- Name of the db to use for this profile --> <!--<param name="dbname" value="share_presence"/>--> <!--<param name="presence-hosts" value="$${domain}"/>--> <!--<param name="force-register-domain" value="$${domain}"/>--> for presence. --> <!-- Name of the db to use for this profile --> <!--<param name="dbname" value="share_presence"/>--> <!--<param name="presence-hosts" value="$${domain}"/>--> <!--<param name="force-register-domain" value="$${domain}"/>--> <!--all inbound reg will stored in the db using this domain --> <!--<param name="force-register-db-domain" value="$${domain}"/>--> <!-- ************************************************* --> <!--<param name="aggressive-nat-detection" value="true"/>--> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <!-- DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS! --> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <!--<param name="enable-3pcc" value="true"/>--> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="$${external_ssl_enable}"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) --> <param name="tls-sip-port" value="$${external_tls_port}"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) --> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 --> <param name="tls-version" value="$${sip_tls_version}"/> </settings>
Regards,
Faisal
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